A communication system with a noise cancellation (NC) assembly providing adaptive or dynamic noise cancellation. The NC assembly includes a localizer module determining, during a communication session (active speaking or during idle times), a location of the active talker. The NC assembly includes a beam generator forming a beam in the determined direction of the active talker to enhance the active talker speech. Once the NC assembly has determined the position of the active talker, the NC assembly assigns a microphone of the microphone array or generated beam in that active direction to be the “active signal” source. The NC assembly assigns a second microphone or beam to be the noise source for NC purposes, and this source may be selected to be in acoustic shadow of the first microphone used as the active signal source or may be the farthest away in its position from the active talker's position.

Microphone arrays comprise several microphone capsules, the outputs of which being electronically combined for directional recording of sound. The directional and frequency properties of the microphone array depend on the number and positions of the microphone array. In order to obtain the smallest possible microphone array with only few microphone capsules, which, however, has an essentially uniform directional and frequency dependence over a speech frequency range, is scalable and robust against small incorrect positioning of the capsules, fifteen or twenty-one microphone capsules (K.sub.15,11-K.sub.15,35, K.sub.21,11-K.sub.21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 120° against each other and forming the grid of equilateral triangles.

Spatial audio signals can include audio objects that can be respectively encoded and rendered at each of multiple different depths. In an example, a method for encoding a spatial audio signal can include receiving audio scene information from an audio capture source in an environment, and receiving a depth characteristic of a first object in the environment. The depth characteristic can be determined using information from a depth sensor. A correlation can be identified between at least a portion of the audio scene information and the first object. The spatial audio signal can be encoded using the portion of the audio scene and the depth characteristic of the first object.

A system and method for determining an in-ear status of wearable audio devices includes determining that the devices are each proximate a portion of a body of a user. Each device determines its orientation; if both orientations correspond to ears of the user, the devices output corresponding audio.

A plurality of modal parameters is estimated for a set of discrete locations within a local area using a set of room impulse responses for the set of discrete locations. The local area includes an audio system. A plurality of acoustic model parameters is generated by fitting the plurality of modal parameters to an acoustic model that accounts for a physical geometry of the local area. The plurality of acoustic model parameters and an indication about the acoustic model are stored for later use by the audio system. The audio system estimates the set of acoustic parameters using the plurality of acoustic model parameters and the indication about the acoustic model, and presents audio content using the set of acoustic parameters.

The speaker array includes a first speaker 1, a second speaker 2, and a local sound emission structure 3. The first transmission portion 311 and the second transmission portion 312 are arranged in such a manner that the distance between the center position of the first transmission portion 311 and the center position of the second transmission portion 312 is smaller than the distance between the center position of the first speaker 1 and the center position of the second speaker 2, in order to generate sound sources that are arranged at an interval narrower than the interval between the first speaker 1 and the second speaker 2.

A beamforming microphone system capable of reducing a processing load caused by calculation of directivity and flexibly setting an area where sound is picked up. A beamforming microphone system according to the present invention includes: a plurality of microphone units; a signal processing unit that processes a sound pickup signal from each microphone unit at each predetermined time; and a storage that associates, for each sound pickup area, sound pickup area information with individual-sound-pickup-area position information freely set within the sound pickup area and stores the information. The signal processing unit includes a position information identification unit that identifies sound source position information, a signal generation unit that generates the sound signal corresponding to the sound from the sound source position, and a channel assignment unit that assigns an output channel, based on the individual-sound-pickup-area position information on the individual sound pickup area to which the sound source position belongs.

A method of detecting passage by an individual carrying a mobile device past an impediment operably connected to an access control is provided. The method including: detecting positional data of a mobile device; transmitting an access request from the mobile device to an access control operably connected to an impediment; detecting at least one of a sound from the impediment and a motion of the impediment; and determining that an individual carrying the mobile device has moved past the impediment in response to at least one of the sound from the impediment, the motion of the impediment, and the positional data of the mobile device.

Embodiments of the disclosure provide methods and apparatuses processing audio data. The method can include: acquiring audio data by an audio capturing device, determining feature information of an enclosure in which the audio capturing device is located, and reverberating the feature information into the audio data.

The present application provides an intelligent device, an intelligent speaker, and a method and system for controlling the same. The intelligent device includes a first sound detection module configured to detect a first sound signal directly reaching the first sound detection module; an angle determination module configured to determine a time difference between the receiving time of the first sound signal and the receiving time of the second sound signal, and determine a relative angle between the intelligent device and the intelligent speaker based on a distance between the first sound detection module and the second sound detection module and the time difference; and a transmitting module configured to transmit a notification message containing the relative angle to the intelligent speaker, so that the intelligent speaker directionally transmits a sound to the intelligent device based on the relative angle. Directional sounding based on relative angle calculation is realized.